Search Results for "interactive c"
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Showing 15 open source projects for "interactive c"
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Siproxd is a proxy/masquerading daemon for the SIP protocol. It allows SIP clients (softphones & hardphones) to work behind an IP masquerading firewall or router.
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OpenSIPS (former OpenSER) is an GPL implementation of a multi-functionality SIP Server that targets to deliver a high-level technical solution (performance, security and quality) to be used in professional SIP server platforms. IMPORTANT: this is no longer the main hosting for the project. This was moved on GITHUB - https://github.com/OpenSIPS/opensips
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SIPSL (SIP Service Layer) is a Programmable Session Border Controller (SBC) and SIP B2BUA, as defined in RFC3261. Written in c++ and multithreaded. The Application logic can be implemented by extending the application class and implementing the call backs.
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GNU Gatekeeper (GnuGk)
H.323 Gatekeeper for VoIP and videconferencing
The project has moved! Please find current versions at https://www.gnugk.org/ The GNU Gatekeeper (GnuGk) is a full featured H.323 gatekeeper under GPL license. It supports VoIP and videoconferencing and includes Radius and database support and many call routing functions. The project has moved! Please find current versions at https://www.gnugk.org/ -
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TurnServer is a implementation of Traversal Using Relay around NAT (TURN) protocol. This protocol allows a client to obtain IP addresses and ports from such a relay.
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VoIP Honey project provides a set of tools for building an entire honeynet, thus includes honeywall and honeypot emulating VoIP environments such as Asterisk PBX or OpenSer with fully configurable connections. Voip Honey runs on GNU/Linux and Windows Systems. It can be compiled for Mac OSX as well.
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skiputler makes your Topcom "Butler 4012 USB VoIP" phone work on Linux and Mac OS X. It's a userspace driver which connects the phone to Skype.
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KAMAILIO (OpenSER) - robust, secure and scalable Open Source (GPL) SIP (RFC3261) server implementation with large features set (over 90 extension modules). As of May 2009, source code is hosted by GIT repository at http://sip-router.org
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MongoDB Atlas runs apps anywhereMongoDB Atlas gives you the freedom to build and run modern applications anywhere—across AWS, Azure, and Google Cloud. With global availability in over 115 regions, Atlas lets you deploy close to your users, meet compliance needs, and scale with confidence across any geography.
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This project is aim to develop module named app_avi.so for asterisk to add the ability of play avi file.
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The goal of the project is to create a high-performance, open-source and standards-compliant implementation of a Home-Subscriber-Server (HSS) for use in a IMS context.
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Simu server project is aimed for IP telephony (VoIP) between classical SIP phones and Flash streaming applications.
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A set of scripts to allow sending and receiving of fax documents on HylaFAX using res_fax and res_fax_digium of Asterisk
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The Asterisk .NET library consists of a set of C# classes that allow you to easily build applications that interact with an Asterisk PBX Server (1.0/1.2/1.4 version). Both FastAGI and Manager API supported. .NET/Mono compatible.
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A Simple Middlebox Configuration Protocol (RFC 4540) implementation for Linux. SIMCO is a signaling protocol that can be used by applications (such as SIP B2BUAs) to dynamically control firewalls and Network Address Translators (NATs).
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